GLES2 randomly does not support the transpose parameter in matrix
uniform calls. So we have to do this manually. Sure it was worth to
mutilate the standard just so all these shitty SoC vendors can safe 3
lines of code.
(Obviously trying to handle all of GLES2 to GL 4.x in a single codebase
was a mistake.)
GLES2 shaders do not have line continuation characters. Abuse the
HAVE_ARRAYS define to exclude code which uses arrays, and which also
happens to cover all code that defines multi-line macros. (So yes, this
is a hack.)
Remove coded_width and coded_height. This was originally added in commit
fd7dde40, when BITMAPINFOHEADER was killed. The separate fields became
redundant in commit e68f4be1. Remove them (nothing passed to the
decoders actually changes with _this_ commit).
Basically abuse the style override mechanism meant for ASS
(mp_ass_set_style()) to update text subtitle styling at runtime too.
This even has the advantage that the style will be overridden, even if
the text subtitle converted (like sd_lavc_conv.c) dares to add a fixed
style in the styles section.
Probably helps with #1622.
This is essentially what it is, and it's a useful for windowing or
downscaling. For upscaling we already have bilinear, no need to cause
extra confusion between biliner and bilinear_slow.
Also made it a bit more well-behaved.
These are EWA-based versions of the keys B/C splines, of which mitchell
is already a member. They are slightly softer and slightly sharper than
mitchell, respectively.
Very easy to define in terms of things we already have.
mitchell, hermite and catmull_rom are all B/C splines and can share the
code which was already written for mitchell. This just redefines them in
terms of that.
This adds a small check for candidates that could potentially be inside
the radius, but aren't necessarily. This speeds up performance by a
negligible amount on my hardware, but it's mainly a prerequisite for a
further change (using a larger true radius).
This echanges the two events hForceFeed/hFeedDone for hResume. This
like the last commit makes things more deterministic.
Importantly, the forcefeed is only done if there is not already a full
buffer yet to be played by the device. This should fix some of the
problems with exclusive mode.
This commit also removes the necessity to have a proxy to the
AudioClient object in the main thread.
fixes#1529
This makes things a bit more deterministic. It ensures that the audio
thread isn't doing anything between IAudioClient_Stop(),
IAudioClient_Reset() and setting the sample_count to 0.
Buffer overfilling on resume is still a problem in exclusive mode (see
next commit).
Previously, this was based on some arbitrary range 1-100, cut off for
no particular reason, and also defined in such a way that higher values
= *less* smoothness. Since it wasn't multiplied by e in the code, the
default had to be 10*e = 28.8539...
Now, it's sane: 1.0 = default, higher = blurrier.
This filter isn't supposed to have a second parameter in the first
place, all literature only uses a single parameter alpha in both places.
The second parameter doesn't even do anything other than adding a
constant factor, which is normalized by the LUT calculation either way.
This is done mainly for consistency, since all of the EWA filters share
similar properties and it's important to distinguish them for
documentation purposes.
No point in duplicating this check all over the place. No point in
really having it in the first place, to be perfectly honest, j1 should
not be THAT badly behaved.
A while ago, we made mpv output the native number of channels by
default, instead of stereo. I assumed this was not wanted for encode
mode.
This commit keeps the assumption, but allows setting the number of audio
output channels at all, instead of always forcing stereo. (Pretty much a
hack.)
The original filter window was design for a radius based on the true
zero, but we always cut it off at our selection of radius either way (by
necessity, due to the square matrix we sample from).
This window is tweaked from the original (true radius) to our actual
cut-off radius, and hence improves the result in a few edge cases. The
main win is the reduction of code complexity, since we no longer need to
know what the true radius actually is.
Commit f54220d9 attempted to improve this, but it got worse. Now there
was a crash when ytdl_hook.lua added external tracks. This happened
because close_unused_demuxers() assumed that sources[0] was the main
demuxer (so that it didn't close it). This assumption failed, because
the ytdl script can add external tracks before the main file is loaded.
The easy fix would have been to check for master_demuxer, and not i==0.
But instead give up on the old idea, make some stricter assumptions how
demuxers and external tracks map, and simplify the code.
Do timeline building (scanning & opening reference files for ordered
chapters, and more) in a thread. As a result, this process can actually
be stopped without having to kill the player.
This is pretty simple: just reuse the demuxer opening thread. We have
to give up on the idea that open_demux_reentrant() is reusable, though.
(Althoughthe timeline readers still need some fixes before they react to
the quit request.)
These functions do blocking work on a separate thread, but wait until
they return. So they are not async or non-blocking. But they do react to
user-input and client API accesses, which makes them reentrant.
Check the scanf() return value, and don't continue if it doesn't find
both numbers (can happen with GLES 1.0). Also, some implementations can
return NULL from glGetString() if something is "broken".
This is a variation of ewa_lanczos that is sinc-windowed instead of
jinc-windowed. Results are pretty similar, but the logic is simpler.
This could potentially replace the ugly ewa_lanczos code.
It's hard to tell, but from comparing stills I think this one has
slightly less ringing than regular ewa_lanczos.
I've reworked pretty much all the logic to correspond to what the theory
actually describes. With this commit, playback is wonderfully smooth on
my machine.
Some of the hacks were not applied if the file format was forced. Commit
37a0c914 moved them to a table, which is checked with normal probing
only.
Fixes#1612 (DVD forces mpeg, which in turn has to export native stream
IDs specifically).
Do some code restructuring on the way. For example, the probescore can
simply be set to the correct initial value, instead of checking whether
it was set at all.
Starting with waf 1.8.6 (in Python 3), the hcode variable isn't a
string, but a byte string.
This commit adds the solution proposed in the upstream waf bug report:
https://code.google.com/p/waf/issues/detail?id=1535
It seems a bit overly verbose, but on the other hand, this solution has
the chance of being most correct/compatible.
Fixes#1604.
